| /* |
| * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #include "modules/rtp_rtcp/source/receive_statistics_impl.h" |
| |
| #include <cmath> |
| #include <cstdlib> |
| #include <memory> |
| #include <utility> |
| #include <vector> |
| |
| #include "api/units/time_delta.h" |
| #include "modules/remote_bitrate_estimator/test/bwe_test_logging.h" |
| #include "modules/rtp_rtcp/source/rtcp_packet/report_block.h" |
| #include "modules/rtp_rtcp/source/rtp_packet_received.h" |
| #include "modules/rtp_rtcp/source/rtp_rtcp_config.h" |
| #include "modules/rtp_rtcp/source/time_util.h" |
| #include "rtc_base/logging.h" |
| #include "rtc_base/time_utils.h" |
| #include "system_wrappers/include/clock.h" |
| |
| namespace webrtc { |
| namespace { |
| constexpr int64_t kStatisticsTimeoutMs = 8000; |
| constexpr int64_t kStatisticsProcessIntervalMs = 1000; |
| } // namespace |
| |
| StreamStatistician::~StreamStatistician() {} |
| |
| StreamStatisticianImpl::StreamStatisticianImpl(uint32_t ssrc, Clock* clock, |
| int max_reordering_threshold) |
| : ssrc_(ssrc), |
| clock_(clock), |
| delta_internal_unix_epoch_ms_(clock_->CurrentNtpInMilliseconds() - |
| clock_->TimeInMilliseconds() - |
| rtc::kNtpJan1970Millisecs), |
| incoming_bitrate_(kStatisticsProcessIntervalMs, |
| RateStatistics::kBpsScale), |
| max_reordering_threshold_(max_reordering_threshold), |
| enable_retransmit_detection_(false), |
| cumulative_loss_is_capped_(false), |
| jitter_q4_(0), |
| cumulative_loss_(0), |
| cumulative_loss_rtcp_offset_(0), |
| last_receive_time_ms_(0), |
| last_received_timestamp_(0), |
| received_seq_first_(-1), |
| received_seq_max_(-1), |
| last_report_cumulative_loss_(0), |
| last_report_seq_max_(-1), |
| last_payload_type_frequency_(0) {} |
| |
| StreamStatisticianImpl::~StreamStatisticianImpl() = default; |
| |
| bool StreamStatisticianImpl::UpdateOutOfOrder(const RtpPacketReceived& packet, |
| int64_t sequence_number, |
| int64_t now_ms) { |
| // Check if `packet` is second packet of a stream restart. |
| if (received_seq_out_of_order_) { |
| // Count the previous packet as a received; it was postponed below. |
| --cumulative_loss_; |
| |
| uint16_t expected_sequence_number = *received_seq_out_of_order_ + 1; |
| received_seq_out_of_order_ = absl::nullopt; |
| if (packet.SequenceNumber() == expected_sequence_number) { |
| // Ignore sequence number gap caused by stream restart for packet loss |
| // calculation, by setting received_seq_max_ to the sequence number just |
| // before the out-of-order seqno. This gives a net zero change of |
| // `cumulative_loss_`, for the two packets interpreted as a stream reset. |
| // |
| // Fraction loss for the next report may get a bit off, since we don't |
| // update last_report_seq_max_ and last_report_cumulative_loss_ in a |
| // consistent way. |
| last_report_seq_max_ = sequence_number - 2; |
| received_seq_max_ = sequence_number - 2; |
| return false; |
| } |
| } |
| |
| if (std::abs(sequence_number - received_seq_max_) > |
| max_reordering_threshold_) { |
| // Sequence number gap looks too large, wait until next packet to check |
| // for a stream restart. |
| received_seq_out_of_order_ = packet.SequenceNumber(); |
| // Postpone counting this as a received packet until we know how to update |
| // `received_seq_max_`, otherwise we temporarily decrement |
| // `cumulative_loss_`. The |
| // ReceiveStatisticsTest.StreamRestartDoesntCountAsLoss test expects |
| // `cumulative_loss_` to be unchanged by the reception of the first packet |
| // after stream reset. |
| ++cumulative_loss_; |
| return true; |
| } |
| |
| if (sequence_number > received_seq_max_) |
| return false; |
| |
| // Old out of order packet, may be retransmit. |
| if (enable_retransmit_detection_ && IsRetransmitOfOldPacket(packet, now_ms)) |
| receive_counters_.retransmitted.AddPacket(packet); |
| return true; |
| } |
| |
| void StreamStatisticianImpl::UpdateCounters(const RtpPacketReceived& packet) { |
| RTC_DCHECK_EQ(ssrc_, packet.Ssrc()); |
| int64_t now_ms = clock_->TimeInMilliseconds(); |
| |
| incoming_bitrate_.Update(packet.size(), now_ms); |
| receive_counters_.last_packet_received_timestamp_ms = now_ms; |
| receive_counters_.transmitted.AddPacket(packet); |
| --cumulative_loss_; |
| |
| int64_t sequence_number = |
| seq_unwrapper_.UnwrapWithoutUpdate(packet.SequenceNumber()); |
| |
| if (!ReceivedRtpPacket()) { |
| received_seq_first_ = sequence_number; |
| last_report_seq_max_ = sequence_number - 1; |
| received_seq_max_ = sequence_number - 1; |
| receive_counters_.first_packet_time_ms = now_ms; |
| } else if (UpdateOutOfOrder(packet, sequence_number, now_ms)) { |
| return; |
| } |
| // In order packet. |
| cumulative_loss_ += sequence_number - received_seq_max_; |
| received_seq_max_ = sequence_number; |
| seq_unwrapper_.UpdateLast(sequence_number); |
| |
| // If new time stamp and more than one in-order packet received, calculate |
| // new jitter statistics. |
| if (packet.Timestamp() != last_received_timestamp_ && |
| (receive_counters_.transmitted.packets - |
| receive_counters_.retransmitted.packets) > 1) { |
| UpdateJitter(packet, now_ms); |
| } |
| last_received_timestamp_ = packet.Timestamp(); |
| last_receive_time_ms_ = now_ms; |
| } |
| |
| void StreamStatisticianImpl::UpdateJitter(const RtpPacketReceived& packet, |
| int64_t receive_time_ms) { |
| int64_t receive_diff_ms = receive_time_ms - last_receive_time_ms_; |
| RTC_DCHECK_GE(receive_diff_ms, 0); |
| uint32_t receive_diff_rtp = static_cast<uint32_t>( |
| (receive_diff_ms * packet.payload_type_frequency()) / 1000); |
| int32_t time_diff_samples = |
| receive_diff_rtp - (packet.Timestamp() - last_received_timestamp_); |
| |
| time_diff_samples = std::abs(time_diff_samples); |
| |
| ReviseFrequencyAndJitter(packet.payload_type_frequency()); |
| |
| // lib_jingle sometimes deliver crazy jumps in TS for the same stream. |
| // If this happens, don't update jitter value. Use 5 secs video frequency |
| // as the threshold. |
| if (time_diff_samples < 450000) { |
| // Note we calculate in Q4 to avoid using float. |
| int32_t jitter_diff_q4 = (time_diff_samples << 4) - jitter_q4_; |
| jitter_q4_ += ((jitter_diff_q4 + 8) >> 4); |
| } |
| } |
| |
| void StreamStatisticianImpl::ReviseFrequencyAndJitter( |
| int payload_type_frequency) { |
| if (payload_type_frequency == last_payload_type_frequency_) { |
| return; |
| } |
| |
| if (payload_type_frequency != 0) { |
| if (last_payload_type_frequency_ != 0) { |
| // Value in "jitter_q4_" variable is a number of samples. |
| // I.e. jitter = timestamp (ms) * frequency (kHz). |
| // Since the frequency has changed we have to update the number of samples |
| // accordingly. The new value should rely on a new frequency. |
| |
| // If we don't do such procedure we end up with the number of samples that |
| // cannot be converted into milliseconds correctly |
| // (i.e. jitter_ms = jitter_q4_ >> 4 / (payload_type_frequency / 1000)). |
| // In such case, the number of samples has a "mix". |
| |
| // Doing so we pretend that everything prior and including the current |
| // packet were computed on packet's frequency. |
| jitter_q4_ = static_cast<int>(static_cast<uint64_t>(jitter_q4_) * |
| payload_type_frequency / |
| last_payload_type_frequency_); |
| } |
| // If last_payload_type_frequency_ is not present, the jitter_q4_ |
| // variable has its initial value. |
| |
| // Keep last_payload_type_frequency_ up to date and non-zero (set). |
| last_payload_type_frequency_ = payload_type_frequency; |
| } |
| } |
| |
| void StreamStatisticianImpl::SetMaxReorderingThreshold( |
| int max_reordering_threshold) { |
| max_reordering_threshold_ = max_reordering_threshold; |
| } |
| |
| void StreamStatisticianImpl::EnableRetransmitDetection(bool enable) { |
| enable_retransmit_detection_ = enable; |
| } |
| |
| RtpReceiveStats StreamStatisticianImpl::GetStats() const { |
| RtpReceiveStats stats; |
| stats.packets_lost = cumulative_loss_; |
| // Note: internal jitter value is in Q4 and needs to be scaled by 1/16. |
| stats.jitter = jitter_q4_ >> 4; |
| if (last_payload_type_frequency_ > 0) { |
| // Divide value in fractional seconds by frequency to get jitter in |
| // fractional seconds. |
| stats.interarrival_jitter = |
| webrtc::TimeDelta::Seconds(stats.jitter) / last_payload_type_frequency_; |
| } |
| if (receive_counters_.last_packet_received_timestamp_ms.has_value()) { |
| stats.last_packet_received_timestamp_ms = |
| *receive_counters_.last_packet_received_timestamp_ms + |
| delta_internal_unix_epoch_ms_; |
| } |
| stats.packet_counter = receive_counters_.transmitted; |
| return stats; |
| } |
| |
| void StreamStatisticianImpl::MaybeAppendReportBlockAndReset( |
| std::vector<rtcp::ReportBlock>& report_blocks) { |
| int64_t now_ms = clock_->TimeInMilliseconds(); |
| if (now_ms - last_receive_time_ms_ >= kStatisticsTimeoutMs) { |
| // Not active. |
| return; |
| } |
| if (!ReceivedRtpPacket()) { |
| return; |
| } |
| |
| report_blocks.emplace_back(); |
| rtcp::ReportBlock& stats = report_blocks.back(); |
| stats.SetMediaSsrc(ssrc_); |
| // Calculate fraction lost. |
| int64_t exp_since_last = received_seq_max_ - last_report_seq_max_; |
| RTC_DCHECK_GE(exp_since_last, 0); |
| |
| int32_t lost_since_last = cumulative_loss_ - last_report_cumulative_loss_; |
| if (exp_since_last > 0 && lost_since_last > 0) { |
| // Scale 0 to 255, where 255 is 100% loss. |
| stats.SetFractionLost(255 * lost_since_last / exp_since_last); |
| } |
| |
| int packets_lost = cumulative_loss_ + cumulative_loss_rtcp_offset_; |
| if (packets_lost < 0) { |
| // Clamp to zero. Work around to accomodate for senders that misbehave with |
| // negative cumulative loss. |
| packets_lost = 0; |
| cumulative_loss_rtcp_offset_ = -cumulative_loss_; |
| } |
| if (packets_lost > 0x7fffff) { |
| // Packets lost is a 24 bit signed field, and thus should be clamped, as |
| // described in https://datatracker.ietf.org/doc/html/rfc3550#appendix-A.3 |
| if (!cumulative_loss_is_capped_) { |
| cumulative_loss_is_capped_ = true; |
| RTC_LOG(LS_WARNING) << "Cumulative loss reached maximum value for ssrc " |
| << ssrc_; |
| } |
| packets_lost = 0x7fffff; |
| } |
| stats.SetCumulativeLost(packets_lost); |
| stats.SetExtHighestSeqNum(received_seq_max_); |
| // Note: internal jitter value is in Q4 and needs to be scaled by 1/16. |
| stats.SetJitter(jitter_q4_ >> 4); |
| |
| // Only for report blocks in RTCP SR and RR. |
| last_report_cumulative_loss_ = cumulative_loss_; |
| last_report_seq_max_ = received_seq_max_; |
| BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "cumulative_loss_pkts", now_ms, |
| cumulative_loss_, ssrc_); |
| BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "received_seq_max_pkts", now_ms, |
| (received_seq_max_ - received_seq_first_), |
| ssrc_); |
| } |
| |
| absl::optional<int> StreamStatisticianImpl::GetFractionLostInPercent() const { |
| if (!ReceivedRtpPacket()) { |
| return absl::nullopt; |
| } |
| int64_t expected_packets = 1 + received_seq_max_ - received_seq_first_; |
| if (expected_packets <= 0) { |
| return absl::nullopt; |
| } |
| if (cumulative_loss_ <= 0) { |
| return 0; |
| } |
| return 100 * static_cast<int64_t>(cumulative_loss_) / expected_packets; |
| } |
| |
| StreamDataCounters StreamStatisticianImpl::GetReceiveStreamDataCounters() |
| const { |
| return receive_counters_; |
| } |
| |
| uint32_t StreamStatisticianImpl::BitrateReceived() const { |
| return incoming_bitrate_.Rate(clock_->TimeInMilliseconds()).value_or(0); |
| } |
| |
| bool StreamStatisticianImpl::IsRetransmitOfOldPacket( |
| const RtpPacketReceived& packet, |
| int64_t now_ms) const { |
| uint32_t frequency_khz = packet.payload_type_frequency() / 1000; |
| RTC_DCHECK_GT(frequency_khz, 0); |
| |
| int64_t time_diff_ms = now_ms - last_receive_time_ms_; |
| |
| // Diff in time stamp since last received in order. |
| uint32_t timestamp_diff = packet.Timestamp() - last_received_timestamp_; |
| uint32_t rtp_time_stamp_diff_ms = timestamp_diff / frequency_khz; |
| |
| int64_t max_delay_ms = 0; |
| |
| // Jitter standard deviation in samples. |
| float jitter_std = std::sqrt(static_cast<float>(jitter_q4_ >> 4)); |
| |
| // 2 times the standard deviation => 95% confidence. |
| // And transform to milliseconds by dividing by the frequency in kHz. |
| max_delay_ms = static_cast<int64_t>((2 * jitter_std) / frequency_khz); |
| |
| // Min max_delay_ms is 1. |
| if (max_delay_ms == 0) { |
| max_delay_ms = 1; |
| } |
| return time_diff_ms > rtp_time_stamp_diff_ms + max_delay_ms; |
| } |
| |
| std::unique_ptr<ReceiveStatistics> ReceiveStatistics::Create(Clock* clock) { |
| return std::make_unique<ReceiveStatisticsLocked>( |
| clock, [](uint32_t ssrc, Clock* clock, int max_reordering_threshold) { |
| return std::make_unique<StreamStatisticianLocked>( |
| ssrc, clock, max_reordering_threshold); |
| }); |
| } |
| |
| std::unique_ptr<ReceiveStatistics> ReceiveStatistics::CreateThreadCompatible( |
| Clock* clock) { |
| return std::make_unique<ReceiveStatisticsImpl>( |
| clock, [](uint32_t ssrc, Clock* clock, int max_reordering_threshold) { |
| return std::make_unique<StreamStatisticianImpl>( |
| ssrc, clock, max_reordering_threshold); |
| }); |
| } |
| |
| ReceiveStatisticsImpl::ReceiveStatisticsImpl( |
| Clock* clock, |
| std::function<std::unique_ptr<StreamStatisticianImplInterface>( |
| uint32_t ssrc, |
| Clock* clock, |
| int max_reordering_threshold)> stream_statistician_factory) |
| : clock_(clock), |
| stream_statistician_factory_(std::move(stream_statistician_factory)), |
| last_returned_ssrc_idx_(0), |
| max_reordering_threshold_(kDefaultMaxReorderingThreshold) {} |
| |
| void ReceiveStatisticsImpl::OnRtpPacket(const RtpPacketReceived& packet) { |
| // StreamStatisticianImpl instance is created once and only destroyed when |
| // this whole ReceiveStatisticsImpl is destroyed. StreamStatisticianImpl has |
| // it's own locking so don't hold receive_statistics_lock_ (potential |
| // deadlock). |
| GetOrCreateStatistician(packet.Ssrc())->UpdateCounters(packet); |
| } |
| |
| StreamStatistician* ReceiveStatisticsImpl::GetStatistician( |
| uint32_t ssrc) const { |
| const auto& it = statisticians_.find(ssrc); |
| if (it == statisticians_.end()) |
| return nullptr; |
| return it->second.get(); |
| } |
| |
| StreamStatisticianImplInterface* ReceiveStatisticsImpl::GetOrCreateStatistician( |
| uint32_t ssrc) { |
| std::unique_ptr<StreamStatisticianImplInterface>& impl = statisticians_[ssrc]; |
| if (impl == nullptr) { // new element |
| impl = |
| stream_statistician_factory_(ssrc, clock_, max_reordering_threshold_); |
| all_ssrcs_.push_back(ssrc); |
| } |
| return impl.get(); |
| } |
| |
| void ReceiveStatisticsImpl::SetMaxReorderingThreshold( |
| int max_reordering_threshold) { |
| max_reordering_threshold_ = max_reordering_threshold; |
| for (auto& statistician : statisticians_) { |
| statistician.second->SetMaxReorderingThreshold(max_reordering_threshold); |
| } |
| } |
| |
| void ReceiveStatisticsImpl::SetMaxReorderingThreshold( |
| uint32_t ssrc, |
| int max_reordering_threshold) { |
| GetOrCreateStatistician(ssrc)->SetMaxReorderingThreshold( |
| max_reordering_threshold); |
| } |
| |
| void ReceiveStatisticsImpl::EnableRetransmitDetection(uint32_t ssrc, |
| bool enable) { |
| GetOrCreateStatistician(ssrc)->EnableRetransmitDetection(enable); |
| } |
| |
| std::vector<rtcp::ReportBlock> ReceiveStatisticsImpl::RtcpReportBlocks( |
| size_t max_blocks) { |
| std::vector<rtcp::ReportBlock> result; |
| result.reserve(std::min(max_blocks, all_ssrcs_.size())); |
| |
| size_t ssrc_idx = 0; |
| for (size_t i = 0; i < all_ssrcs_.size() && result.size() < max_blocks; ++i) { |
| ssrc_idx = (last_returned_ssrc_idx_ + i + 1) % all_ssrcs_.size(); |
| const uint32_t media_ssrc = all_ssrcs_[ssrc_idx]; |
| auto statistician_it = statisticians_.find(media_ssrc); |
| RTC_DCHECK(statistician_it != statisticians_.end()); |
| statistician_it->second->MaybeAppendReportBlockAndReset(result); |
| } |
| last_returned_ssrc_idx_ = ssrc_idx; |
| return result; |
| } |
| |
| } // namespace webrtc |